similar to: what is the best codec for low bandwidth? for quality?

I've just gotten off a skype conference call and it pisses me off that the quality of skype is higher than my asterisk calls. Is there such a thing as a super high bandwidth codec? In a situation that you have the bandwidth to share is there something that I can use for important calls when the situation warrants it? TIA, Dean -------------- next part -------------- An

Hello, everybody: I'm developing an application using Asterisk and a TDM-400 card. I understand the concept of the difference between GSM and WAV files when using Asterisk, but I'm not happy with the sound quality with the GSM compression. It's merely *acceptable* for a telephone call, but for anything else, it leaves something to be desired. Case in point -- if you compare the

Decided to upgrade to 7.x. It's been a good ride 6.x, but you're living in the past. Everything is backed up, just waiting on my new 4TB HD to arrive for a fresh install. See ya'll on the other side!!

Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls,

Hi ya'll, I built a new CentOS 5.1 server for a client, housing a Lotus Notes / Domino server, and various other virtualized IBM software server guests, and soon will have to physically move that server to another distant location. My question is that I will need secure access to those servers via X, not just the C/L terminal. What do you recommend for a good secure CentOS program which

Hi I am having a problem with voice quality getting bad after about 10-15 minutes on a call. The call starts out fine but gets very chopy after 10-15 minutes. It seems like a buffer problem, I am using Asterisk with a Voicetronix Openlin4 card. Could someone shed some light on this?

Group, Just want to share with the group my recent findings regarding CODECs/Vocoders and the effect it has had on voice quality and the intermittent noise and breakup problem I have which I mentioned in a previous emailing with the u-law CODEC. Calls again are placed through a SIP phone to a TDM400P to the PSTN. A good reference on the reasoning behind the selection of a CODEC was found in the

A while back, I posted a note asking if anyone had any ideas why the /etc/mail/access file was not being parsed or utilized in the efforts to stop spam and junk mail. I just looked over things again, and have still not found any reason why it still permits the TLD's I have listed to pass thru. I also thought perhaps there might be some "upper limit" to the number of entries

On Thu, Jan 10, 2013 at 2:18 PM, Justin Holewinski <justin.holewinski at gmail.com> wrote: > On Thu, Jan 10, 2013 at 1:31 PM, Daniel Berlin <dberlin at dberlin.org> wrote: >> >> >> >> >> I do not think you can legally make clang a system compiler on >> >> Windows without licensing headers/libraries from Microsoft. >> >> So

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

Hello, I'd like to implement some public sip uri's that poeple can call into and get an echo test. Is there a way to force a codec so that users can test various codecs? Something like: echo-test at example.com (negotiates whatever codec, is there a way to figure out what codec was negotiated and tell the user) echo-test-g711 at example.com (forces g711) echo-test-g729 at

blink : It use iLBC also. Voice over IP RTP: A Transport Protocol for Real-Time Applications RFC3550 RCTP: Real Time Control Protocol attribute in Session Description Protocol RFC3605 SRTP: The Secure Real-time Transport Protocol RFC3711 DTMF: Dual-tone Multi-frequency Signaling RFC2833 and inband MWI: Message Summary Event Package RFC3842 Speex and G722: Wide-band Internet Codecs G711, iLBC

Hi, I have a few methodological and implementation questions for ya'll. Thank you in advance for your help. I have a dataset that reflects people's preference choices. I want to see if there's any kind of clustering effect among certain preference choices (e.g. do people who pick choice A also pick choice D). I have a data set that has one record per user ID, per preference choice.

Hi All, I've been playing with GSM and iLBC over low bandwidth connections (central Asterisk box with 2mbps, to ADSL users on 512/256) and both seem to perform well. Based upon what I've read in the archives and at voip-info.org iLBC should perform a little better if packets are lost, than compared to GSM. Do you find this to be true in practice, or is GSM just as robust? Whilst

Hi! As the upstream of my DSL-connection is very slow, I'd like my sip-phones to use iLBC to connect to my *. My gateway provider only allows ulaw. Hence, I'd like to use the follwing setup: SIP-phone <--iLBC--> Asterisk <---ulaw----> PSTN-Gateway I get the following error: "Unable to find a codec translation path from ilbc to ulaw" Setup SIP-phone: disallow=all

I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and they all are able to register and make calls with no problem . My voip carrier supports gsm as well as ilbc .. Server takes calls from sip phones , does call recording in between and forwards to voip carrier . My problem is that half of my softphones use ilbc and rest use gsm and my provider supports both gsm as well as

Greetings all, I'm trying to improve the codec selection on a few of the asterisk boxes we have to keep the g729 licences free for calls from ATAs that don't support anything apart from g711 and g729. GSM seems to offer noticably inferior call quality (at least when using a softphone + decent headphones), but it's about where I want the bitrate to be. I know there are lots of Speex

On Thu, Jan 10, 2013 at 2:59 PM, Daniel Berlin <dberlin at dberlin.org> wrote: > On Thu, Jan 10, 2013 at 2:18 PM, Justin Holewinski > <justin.holewinski at gmail.com> wrote: > > On Thu, Jan 10, 2013 at 1:31 PM, Daniel Berlin <dberlin at dberlin.org> > wrote: > >> > >> >> > >> >> I do not think you can legally make clang a

Hi, I'm having trouble explicitly forcing a codec between sip devices. Am I missing something or is this not really possible? I have a grandstream registering to asterisk, named sip0. Sip0 registers, via sip, to another asterisk box, sip1. When I place a call from the grandstream, it will travel through sip0 to sip1, where it is then placed to the PSTN. Nothing can reinvite, this path is

> -----Original Message----- > From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Mark > LaPierre > Sent: Monday, January 23, 2017 7:30 PM > To: centos at centos.org; Mark LaPierre > Subject: Re: [CentOS] I'm making the change to a new OS > > On 01/22/17 23:12, TE Dukes wrote: > > Decided to upgrade to 7.x. > > > > It's been a good